A speakerphone functions as a "hands-free" telephone wherein a loudspeaker and a microphone perform the typical handset functions of transmitting and receiving speech signals. As a result, the close proximity of a user to the phone is not required.
One problem associated with speakerphones is related to a noise component which results from acoustic coupling between the loudspeaker and the microphone. This undesirable acoustic coupling is commonly known as "room echo". The room echo noise component results from both direct and reflected sound paths from the loudspeaker to the microphone and, unless suppressed, the room echo noise component is audible to the caller. A second echo path, the trunk echo, exists at the near-end analog hybrid coupling device and is caused by an inherent impedance mismatch between the coupling device and the line. The overall effect of these echo noise components is to create a signal loop having positive gain. In order to avoid loop oscillations, the echo noise components must be suppressed or cancelled.
More specifically, the room echo noise component may be represented as the overall linear transfer function from the loudspeaker to to the microphone. The shape and duration of this transfer function are determined in part by the physical structure of the speakerphone, especially the relative positions of the microphone and the loudspeaker, and also by the dimensions, content and sound absorption properties of the room wherein the speakerphone is located. For example, the reverberation time of a typical office is approximately 330 milliseconds. Reverberation time is generally defined as the time required for the echo to decay to -60 dB or less.
One conventional approach to reducing room echo signal components is to require that the microphone and loudspeaker be physically separated by placing them in separate enclosures. Another conventional approach is to require an operator to speak into a handset while simultaneously inputting the speech through the microphone. In that the handset generated speech signal is, ideally, free of room echo components, circuitry with the speakerphone is enabled to identify and compensate for the echo component of the audio signal. Obviously, neither of these conventional approaches represents an optimum solution.
The trunk echo is the sum of all signal reflections due to the impedance mismatch in any two-to-four wire conversion hybrid circuits along the connection path. The strongest reflection occurs at the CBX connection to the trunk and is typically referred to as the near-end hybrid echo. This trunk echo component has a typical duration, including some delay, of 16 milliseconds. A far-end hybrid echo may also exist.
Trunk echoes of relatively long delay time are usually suppressed within the telephone network by Via Net Loss (VNL) and/or echo suppression/cancellation methods. However, domestic terrestrial trunks of up to 1850 airline miles in length are not required to include suppressors and may produce echo delays averaging as much as 35 milliseconds.
In a half-duplex speakerphone, the undesirable effects of such room and trunk echoes may be avoided by enabling only the transmit or the receive signal path at any given time. The signal path control mechanism is typically some form of a voice-operated switch. However, due to the finite switching time between the signal paths a user of a conventional half-duplex speakerphone may experience a degradation of interactive conversation, a limited ability to interrupt the talking party and a clipping of first syllables.
A full-duplex speakerphone ideally provides simultaneous bidirectional conversation and thus does not experience those problems related to signal path switching which occur in a half-duplex speakerphone. Typically an attempt is made to cancel both room and trunk echoes by employing adaptive filter techniques. For example, the signal which is coupled to the loudspeaker is also coupled through an adaptive filter and is thereafter subtracted from the microphone input signal. Ideally the adaptive filter is designed to minimize the difference between the microphone input signal and the output of the adaptive filter. Thus, in the ideal case, any non-echo signal is coupled through the speakerphone while the echo signal component is cancelled.
However, conventional full-duplex speakerphones do not operate in accordance with these ideal characteristics. Conventional full-duplex speakerphones exhibit problems related to the time required to initialize, or adapt, the adaptive filters. Furthermore, in a digital full-duplex speakerphone, error signals related to either A-Law or Mu-Law quantization and linearity errors cause a degradation in performance.